JPH0730432A - Device and method for recording and/or reproducing or transmitting and/or receiving compressed data, and recording medium - Google Patents

Abstract

PURPOSE:To prevent a sampling frequency signal generating circuit from being complicated and hardware size from being expanded by using the same sampling frequency. CONSTITUTION:An input signal is divided at its band by band dividing filters 101, 102 and outputs from the filters 101, 102 are respectively supplied to respective orthogonal transformation circuits 103 to 105. Block size determined by an orthogonal transformation size determining circuit 106 is simultaneously supplied to the circuits 103 to 105 and respective filter outputs are blocked in accordance with the block size and orthogonally transformed. A bit distribution calculating circuit 107 finds out masking quantity in each divided band while considering a critical band and block floating and the number of allocated bits in each band. An adaptive bit allocation encoding circuit 108 requantizes each spectrum data in accordance with the number of bits allocated to each band. Thus one kind of sampling frequency is used.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a compressed data recording and / or reproducing or transmitting and / or receiving apparatus in which a digital audio signal or the like is bit-compressed, a compressed data recording and / or reproducing or transmitting and / or receiving method, and The present invention relates to a recording medium, and more particularly to an apparatus, a method and a recording medium for recording in a compressed mode of a plurality of bit rates.

[0002]

2. Description of the Related Art The applicant of the present invention has previously disclosed a technique for bit-compressing an input digital audio signal and burst-recording a predetermined data amount as a recording unit, for example, Japanese Patent Application No. 2-221364. , Japanese Patent Application No. 2-22136
No. 5, Japanese Patent Application No. 2-222821, Japanese Patent Application No. 2-2228
No. 23 specification, drawings, etc.

This technique uses a magneto-optical disk as a recording medium and records AD (adaptive difference) PCM audio data defined in the audio data format of so-called CD-I (CD-interactive) or CD-ROM XA. For example, 32 sectors of this ADPCM data and several sectors for linking for interleave processing are recorded as a recording unit on the magneto-optical disk in a burst manner.

Several modes can be selected for the ADPCM audio in the recording / reproducing apparatus using the magneto-optical disk, which is, for example, twice as long as the reproducing time of a normal so-called compact disc (CD). Level A with a compression rate and a sampling frequency of 37.8 kHz,
With a compression rate of 4 times, the sampling frequency is 37.8 kHz, level B, and with a compression rate of 8 times, the sampling frequency is 18.
A level C of 9 kHz is specified. That is, for example, in the case of the level B, the digital audio data is compressed to approximately 1/4, and the reproduction time (play time) of the disc recorded in the level B mode is the standard CD format (CD -DA format). This is a smaller disc, standard 1
Since the recording / playback time is about the same as 2 cm,
The device can be miniaturized.

However, the rotation speed of the disk is standard C
Since it is the same as D, for example, in the case of the above level B, compressed data for a reproduction time that is four times that of the predetermined time is obtained. Therefore, for example, the same compressed data is read four times in a time unit such as a sector or a cluster, and only the compressed data for one time is sent to the audio reproduction. Specifically, when scanning (tracking) a spiral recording track,
A reproduction operation is performed in such a form that a track jump is performed to return to the original track position for each rotation, and the same track is repeatedly tracked four times. This means that normal compressed data only needs to be obtained at least once out of, for example, four times of redundant reading, is resistant to errors due to disturbances, etc., and is particularly preferable when applied to a portable small device.

Further, it is considered that a semiconductor memory will be used as a recording medium in the future, and it is desirable to perform additional bit compression in order to further improve the compression efficiency. Specifically, it is the one in which an audio signal is recorded and reproduced using a so-called IC card.
The compressed data that has been bit-compressed is recorded and reproduced on the card.

IC cards and the like using such a semiconductor memory are expected to have an increased recording capacity and a reduced price with the progress of semiconductor technology, but at the initial stage when they are supplied to the market. It is thought that the capacity is low and expensive. Therefore, it is sufficiently conceivable to transfer the contents from another inexpensive and large-capacity recording medium such as the above-mentioned magneto-optical disk to an IC card or the like for frequent rewriting and use. Specifically, for example, of a plurality of songs recorded on the magneto-optical disk, a favorite song is dubbed to an IC card, and when it is no longer needed, it is replaced with another song. By frequently rewriting the contents of the IC card in this manner, various songs can be enjoyed outdoors, etc., with a small number of IC cards held in hand.

[0008]

By the way, at the time of recording / reproducing an audio signal, there are various applications.
The required bandwidth and noise-to-signal characteristics are different. For example, when high quality audio is required, the bandwidth is 1
5 kHz to 20 kHz is required and good signal-to-noise characteristics are also required. A higher bit rate to achieve this is acceptable. Usually, the bit rate is 256 kbps to 64 kbps / channel.

On the other hand, when a voice signal is mainly handled, the bandwidth may be 5 kHz to 7 kHz, and the signal-to-noise characteristic need not be so high. However, in order to lengthen the recording / playback time as much as possible, the bit rate is 64k.
It is required to reduce from bps to several kbps.
It is necessary to provide a recording / reproducing apparatus which can satisfy such a plurality of applications having different required levels and which has an economical load as small as possible. However, when trying to have multiple modes with different bandwidths, until now, it was unavoidable to support multiple sampling frequencies, and the sampling frequency signal generation circuit was complicated and the hardware scale was inevitable. . When the sampling frequency of each mode is different, it is difficult to move information between the modes, and it is desired to write the high bit rate mode information on the large capacity magneto-optical disk to the small capacity IC card in the low bit rate mode. At times, it was necessary to completely cancel the compression mode, restore the signal on the time axis, and then perform compression processing in the low bit rate mode again, so the amount of processing calculation was large and it was difficult in real time.

Next, as the mode becomes lower in bit rate, the sound quality deteriorates due to the reduction of usable bits. When the bandwidth is narrowed and the frequency division width for compression is constant regardless of the frequency, if the 20 kHz band is divided into 32, a low critical bandwidth of 100 Hz is obtained.
On the other hand, the divided bandwidth becomes as wide as 700 Hz, which is wider than the critical band in most of the middle and low frequencies, and the compression efficiency is remarkably lowered. Further, when the bit rate is reduced, if the bit amount is reduced so that only one of the main information and the sub information of the high efficiency code is deviated, the sound quality deteriorates significantly. Therefore, it is necessary to reduce not only the main information but also the sub information.

The present invention has been made in view of the above circumstances. First, when it is desired to have a plurality of bit rate modes, the sampling frequency signal generating circuit is prevented from becoming complicated and the hardware scale being increased. Secondly, when dubbing bit-compressed data from a recording medium such as the magneto-optical disc or optical disc onto another recording medium such as the IC card, or when compressing bit-compressed data from another recording medium such as the IC card. It is an object of the present invention to provide a compressed data recording and / or reproducing apparatus which can be performed with a small amount of calculation when reproducing, and thirdly to prevent deterioration of sound quality in the low bit rate mode as much as possible.

[0012]

A compressed data recording and / or reproducing apparatus, method or recording medium according to the present invention uses a plurality of sampling frequencies regardless of the difference in bit rate of each mode. It is possible to prevent the sampling frequency signal generation circuit from becoming complicated and the hardware scale from increasing in the case of the above.

Further, it is difficult to move information between the modes, which is difficult when the sampling frequency of each mode is different. For example, high bit rate mode information on a large capacity magneto-optical disk can be transferred to a small capacity IC card. When you want to write in the low bit rate mode, it is not necessary to completely cancel the compression mode and return to the signal on the time axis, and you can realize the low bit rate mode compression processing with only additional processing. The increase in the amount of calculation is minimized and real-time processing is possible. Furthermore, when converting from the low bit rate mode to at least the higher bit rate mode, the format conversion, that is, the rearrangement of the encoded data, can be performed.

In the present invention, in order to make the compatibility of compressed data (encoded data) as high as possible, the configurations of the orthogonal transform block sizes are made equal in each bit rate mode, so that the conversion between different bit rate modes can be performed in a timely manner. It can be performed at high speed without restoring the axis signal. Also, as the bit rate becomes lower, the blocks for floating blocks and / or the quantization noise generating blocks that are adjacent to each other in the time axis direction or the frequency axis direction are shared, so that the blocks for block floating and The amount of sub-information such as so-called scale factor and word length required for each quantization noise generation block is reduced. Generally, since the acoustic signal has a high correlation in both the time axis direction and the frequency axis direction, even if the block for the block floating and / or the quantization noise generating block are shared, the influence on the sound quality is small,
In the case of a non-stationary signal in time, the orthogonal transformation block size is made variable to prevent a reduction in compression efficiency. The bits for the reduced sub information can be added to the main information.

Further, in all modes, when the frequency division width for controlling the quantization noise is constant regardless of the frequency, when the 20 kHz band is divided into 32, the lower critical band width is 100 Hz. 700Hz
It becomes very wide, and becomes narrower than the critical band in the middle and low frequencies, resulting in a marked decrease in efficiency. In the present invention, the frequency division width for controlling the quantization noise is selected so that it becomes close to the critical bandwidth, and becomes wider as the frequency becomes higher in at least most of the frequency division bands.

In the case of dubbing bit-compressed data from a recording medium such as the magneto-optical disk to another recording medium such as the IC card, at least the bit expansion is not completely performed, and the dubbing is performed as it is or after additional compression. To do. In the case of additional compression, the signal is not converted at all, the bits are redistributed and requantized on the frequency axis,
Record multiple sub-information in common.

[0017]

According to the compressed data recording and / or reproducing apparatus, method or recording medium according to the present invention, by using one kind of sampling frequency, a sampling frequency signal generating circuit which occurs when there are a plurality of sampling frequencies is provided. It is possible to prevent complication and increase in hardware scale. In addition, information transfer between modes with different bit rates can be easily performed without complicated operations such as sampling frequency conversion, and high bit rate mode information on a large capacity magneto-optical disk is written in a small capacity IC card in a low bit rate mode. When it is desired to perform the above, the low bit rate mode compression processing can be realized only by the additional processing, the increase in the amount of processing calculation can be suppressed to the minimum, and the real-time processing can be performed. Further, according to the present invention, it is possible to prevent deterioration of sound quality in a mode having a low bit rate.

[0018]

Embodiments of the present invention will be described below with reference to the drawings.

First, FIG. 1 is a block circuit diagram showing a schematic configuration of an embodiment of a compressed data recording and / or reproducing apparatus according to the present invention.

In the recording / reproducing apparatus of FIG. 1, two units, a recording / reproducing unit of a magneto-optical disk 1 which is one recording medium and a recording unit of an IC card 2 which is another recording medium, are combined into one system. It is composed by combining. When the signal reproduced by the reproducing system on the side of the magneto-optical disk recording / reproducing unit is recorded by the IC card recording unit, the optical head 53 is moved from the magneto-optical disk 1 of the reproducing system.
The reproduced compressed data (ATC audio data), which has been read by, and is sent to the decoder 71 and subjected to EFM demodulation, deinterleave processing, error correction processing, etc., is sent to the memory 85 of the IC card recording unit, and this memory is 8
5 is subjected to additional processing of a variable bit rate coding processing window by an additional compressor 84 which performs additional processing such as entropy coding, and is recorded in the IC card 2 via the IC card interface circuit 86. In this way, the reproduced compressed data is sent to the recording system in the compressed state before being subjected to the decompression processing by the ATC decoder 73 and recorded in the IC card 2.

By the way, during normal reproduction (for audio listening), a predetermined data amount unit (eg, 3) is intermittently or bursted from the recording medium (magneto-optical disk 1).
Compressed data is read out in 2 sectors + several sectors) and is expanded and converted into a continuous audio signal. During so-called dubbing, the compressed data on the medium is continuously read and sent to the recording system for recording. is doing. This enables high-speed (short-time) dubbing according to the data compression rate.

The specific configuration of FIG. 1 will be described in detail below. In the magneto-optical disk recording / reproducing unit of the compressed data recording and / or reproducing apparatus shown in FIG. 1, the magneto-optical disk 1 rotatably driven by the spindle motor 51 is used as the recording medium. When recording data on the magneto-optical disk 1, for example, the optical head 53
By applying a modulation magnetic field according to the recording data with the magnetic head 54 while irradiating the laser beam with
So-called magnetic field modulation recording is performed, and data is recorded along the recording track of the magneto-optical disk 1. Also during playback,
The recording track of the magneto-optical disk 1 is traced with a laser beam by the optical head 53 to reproduce magneto-optically.

Hereinafter, the recording / reproducing apparatus will be mainly described. The optical head 53 is composed of, for example, a laser light source such as a laser diode, a collimator lens, an objective lens, a polarization beam splitter, an optical component such as a cylindrical lens, and a photodetector having a light receiving portion of a predetermined pattern. The optical head 53 is provided at a position facing the magnetic head 54 through the magneto-optical disk 1. When data is recorded on the magneto-optical disk 1, the magnetic head 54 is driven by a recording system head drive circuit 66 described later to apply a modulation magnetic field according to the recording data, and the optical head 53 is used by the optical head 53.
By irradiating the target track with laser light, thermomagnetic recording is performed by the magnetic field modulation method. Further, the optical head 53 detects the reflected light of the laser light applied to the target track, detects a focus error by, for example, a so-called astigmatism method, and detects a tracking error by, for example, a so-called push-pull method. When reproducing data from the magneto-optical disk 1, the optical head 53 detects the focus error and the tracking error, and at the same time, detects the difference in the polarization angle (Kerr rotation angle) of the reflected light of the laser light from the target track and reproduces it. Generate a signal.

The output of the optical head 53 is supplied to the RF circuit 55. The RF circuit 55 extracts the focus error signal and the tracking error signal from the output of the optical head 53 and supplies them to the servo control circuit 56, and binarizes the reproduction signal to reproduce the reproduction system decoder 7 described later.
Supply to 1.

The servo control circuit 56 is composed of, for example, a focus servo control circuit, a tracking servo control circuit, a spindle motor servo control circuit, a sled servo control circuit, and the like. The focus servo control circuit controls the focus of the optical system of the optical head 53 so that the focus error signal becomes zero. Further, the tracking servo control circuit controls the tracking of the optical system of the optical head 53 so that the tracking error signal becomes zero. Further, the spindle motor servo control circuit controls the spindle motor 51 so as to rotate the magneto-optical disk 1 at a predetermined rotation speed (for example, a constant linear speed). Further, the sled servo control circuit moves the optical head 53 and the magnetic head 54 to the target track position of the magneto-optical disk 1 designated by the system controller 57. The servo control circuit 56 that performs such various control operations sends information indicating the operating state of each unit controlled by the servo control circuit 56 to the system controller 57.

A key input operation section 58 and a display section 59 are connected to the system controller 57. The system controller 57 controls the recording system and the reproducing system in the operation mode designated by the operation input information from the key input operation unit 58. The system controller 7 also uses the optical head 53 and the magnetic head 54 based on the address information in sector units reproduced from the recording track of the magneto-optical disk 1 by the header time, the Q data of the subcode, or the like.
Manages the recording position and the reproducing position on the recording track traced by the. Further system controller 57
Is bit compression mode information in an ATC (Adaptive Transform Coding) encoder 63, which will be described later, which is switched and selected by the key input operation unit 58, and a bit compression mode in reproduction data obtained from the RF circuit 55 via a reproduction system, which will be described later. The bit compression mode is displayed on the display unit 59 based on the information.
And the reproduction time is displayed on the display unit 59 based on the data compression rate in the bit compression mode and the reproduction position information on the recording track.

This reproduction time display is based on data in the bit compression mode for address information (absolute time information) in sector units reproduced from the recording track of the magneto-optical disk 1 by so-called header time or so-called sub-code Q data. By multiplying the reciprocal of the compression rate (for example, 4 in the case of 1/4 compression), the actual time information is obtained and displayed on the display unit 9. Even at the time of recording, when absolute time information is recorded (pre-formatted) in advance on a recording track of a magneto-optical disk or the like, the pre-formatted absolute time information is read to determine the data compression rate. It is also possible to display the current position at the actual recording time by multiplying by the reciprocal.

Next, in the recording system of the recording / reproducing apparatus of this disk recording / reproducing apparatus, the analog audio input signal AIN from the input terminal 60 is passed through the low-pass filter 61 to A
The A / D converter 62 is supplied to the A / D converter 62, which quantizes the analog audio input signal AIN. A /
The digital audio signal obtained from the D converter 62 is supplied to the ATC encoder 63. Further, the digital audio input signal DIN from the input terminal 67 is supplied to the ATC encoder 63 via the digital input interface circuit 68. The ATC encoder 63 is
A bit compression (data compression) process corresponding to various modes in the ATC system shown in Table 1 is performed on digital audio PCM data of a predetermined transfer rate obtained by quantizing the input signal AIN by the A / D converter 62.
The operation mode is designated by the system controller 57. For example, in B mode, the sampling frequency is 44.1 kHz and the bit rate is 64 k
Compressed data (ATC data) of bps and stored in the memory 6
4 is supplied. The data transfer rate in the B mode stereo mode is 1/8 (9.375) of the standard CD-DA format data transfer rate (75 sectors / second).
Sector / second).

[0029]

[Table 1]

In the embodiment shown in FIG. 1, the sampling frequency of the A / D converter 62 is, for example, the standard C mentioned above.
4 which is the sampling frequency of D-DA format
The ATC encoder 13 is fixed at 4.1 kHz.
Also in, it is assumed that the sampling frequency is maintained and bit compression processing is performed. At this time, as the low bit rate mode is set, the signal pass band becomes narrower, and accordingly, the cutoff frequency of the low pass filter 61 is also switched and controlled. That is, the cutoff frequency of the low-pass filter 61 of the A / D converter 62 may be simultaneously switched and controlled according to the compression mode.

Next, in the memory 64, the writing and reading of data are controlled by the system controller 57, the ATC data supplied from the ATC encoder 63 is temporarily stored, and recorded on the disk as needed. It is used as a buffer memory for That is, for example, in the stereo mode of the B mode, the compressed audio data supplied from the ATC encoder 63 has a standard CD-D data transfer rate.
It is reduced to 1/8 of the data transfer rate of A format (75 sectors / second), that is, 9.375 sectors / second, and this compressed data is continuously written in the memory 64. As for the compressed data (ATC data), it is sufficient to record one sector for every eight sectors as described above, but since recording every eight sectors is practically impossible, the continuous sectors as described later are recorded. I try to keep a record. This recording bursts at a data transfer rate (75 sectors / second) same as that of the standard CD-DA format by using a cluster composed of a plurality of predetermined sectors (for example, 32 sectors + several sectors) as a recording unit through a pause period. Is done in a regular manner. That is, in the memory 64, the ATC audio data in the B mode and the stereo mode continuously written at the low transfer rate of 9.375 (= 75/8) sectors / sec corresponding to the bit compression rate is recorded data. The data is read in bursts at the transfer rate of 75 sectors / sec. The overall data transfer rate of the read and recorded data, including the recording pause period, is as low as 9.375 sectors / sec, but within the time of the recording operation performed in a burst manner. The instantaneous data transfer rate in the above is the standard 75 sectors / sec. Therefore, when the disc rotation speed is the same as the standard CD-DA format (constant linear velocity), the CD-
Recording with the same recording density and storage pattern as in the DA format will be performed.

The ATC audio data, that is, the recording data, which is burst-read from the memory 64 at the (instantaneous) transfer rate of 75 sectors / second is recorded by the encoder 65.
Is supplied to. Here, from the memory 64 to the encoder 65
In the data string supplied to the above, the unit of continuous recording in one recording is a cluster composed of a plurality of sectors (for example, 32 sectors) and several sectors for cluster connection arranged in the front and rear positions of the cluster. This cluster connection sector is set longer than the interleave length in the encoder 65 so that the data of other clusters will not be affected even if interleaved.

The encoder 65, regarding the recording data supplied from the memory 64 in bursts as described above,
Encoding processing (parity addition and interleave processing) for error correction, EFM encoding processing, and the like are performed. The recording data encoded by the encoder 65 is supplied to the magnetic head drive circuit 66. The magnetic head drive circuit 66 is connected to the magnetic head 54,
The magnetic head 54 is driven so as to apply the modulation magnetic field according to the recording data to the magneto-optical disk 1.

The system controller 57 controls the memory 64 as described above, and
By this memory control, the recording position is controlled so that the recording data read in burst from the memory 64 is continuously recorded on the recording track of the magneto-optical disk 1.
The recording position is controlled by controlling the recording position of the recording data which is burst-read from the memory 64 by the system controller 57 and outputting a control signal for designating the recording position on the recording track of the magneto-optical disk 1 to the servo control circuit. By feeding 56.

Next, the reproducing system of this magneto-optical disk recording / reproducing unit will be described. This reproducing system is for reproducing the record data continuously recorded on the recording track of the magneto-optical disk 1 by the above-mentioned recording system,
The decoder 71 is provided with a reproduction output obtained by tracing a recording track of the magneto-optical disk 1 with a laser beam by the optical head 53 and binarized and supplied by the RF circuit 55. At this time, not only the magneto-optical disk but also the read-only optical disk same as the compact disk can be read.

The decoder 71 corresponds to the encoder 65 in the recording system described above, and decodes the reproduced output binarized by the RF circuit 55 as described above for error correction and EFM decoding. By performing processing such as processing, the above-described B-mode stereo mode ATC audio data is reproduced at a transfer rate of 75 sectors / second, which is faster than the normal transfer rate in the B-mode stereo mode. The reproduction data obtained by this decoder 71 is
It is supplied to the memory 72.

In the memory 72, writing and reading of data are controlled by the system controller 57, and reproduced data supplied from the decoder 71 at a transfer rate of 75 sectors / second is written in bursts at the transfer rate of 75 sectors / second. Be done. Further, in the memory 72, the reproduction data written in a burst at the transfer rate of 75 sectors / second is the normal 9.37 in the stereo mode of the B mode.
It is continuously read at a transfer rate of 5 sectors / second.

The system controller 57 writes the reproduced data in the memory 72 at a transfer rate of 75 sectors / second, and also writes the reproduced data from the memory 72 in the above 9.37.
Memory control is performed so that data is continuously read at a transfer rate of 5 sectors / second. Further, the system controller 57 performs the above-mentioned memory control on the memory 72, and continuously reproduces the above-mentioned reproduction data written in burst from the memory 72 from the recording track of the magneto-optical disk 1 by the memory control. Controls the playback position. This playback position is controlled by the system controller 5
By controlling the reproduction position of the reproduction data which is read out from the memory 72 in burst by 7 and supplying a control signal for designating the reproduction position on the recording track of the magneto-optical disc 1 or the optical disc 1 to the servo control circuit 56. Done.

Stereo mode ATC audio data of B mode obtained as reproduction data continuously read from the memory 72 at a transfer rate of 9.375 sectors / second is supplied to the ATC decoder 73. The ATC decoder 73 corresponds to the ATC encoder 63 of the recording system, the operation mode is designated by the system controller 57, and, for example, the B mode stereo mode ATC data is expanded eight times (bit expansion). As a result, 16-bit digital audio data is reproduced. The digital audio data from the ATC decoder 73 is supplied to the D / A converter 74.

The D / A converter 74 is the ATC decoder 73.
The digital audio data supplied from the converter is converted into an analog signal to form an analog audio output signal AOUT. The analog audio signal AOUT obtained by the D / A converter 74 is output from the output terminal 76 via the low pass filter 75.

Next, the IC card recording unit of the compressed data recording and / or reproducing apparatus will be described. The analog audio input signal AIN from the input terminal 81 is supplied to the A / D converter 83 via the low pass filter 82 and quantized. The digital audio signal obtained from the A / D converter 62 is sent to an additional compressor 84 that performs so-called entropy encoding, which is a type of variable bit rate encoder, and is subjected to entropy encoding and other processing.
This process is executed while reading / writing data from / to the memory 85. The variable bit rate compression-encoded data from the additional compressor 84 that performs entropy encoding or the like is transmitted via the IC card interface circuit 86 to I
It is recorded on the C card 2. Of course, in the present invention, variable bit rate compression such as entropy coding is not performed, but the size of orthogonal transform is increased, and the block for block floating on the frequency axis having sub information and / or the block for quantization noise generation is used. Recording may be performed at a lower bit rate and a constant bit rate by expanding the frequency width.

Here, compressed data (ATC) from the decoder 71 of the reproducing system of the magneto-optical disk recording / reproducing unit is used.
(Data) is sent to the memory 85 of the IC card recording unit without being expanded. This data transfer is performed by the system controller 57 controlling the memory 85 and the like during so-called high-speed dubbing. The compressed data from the memory 72 may be sent to the memory 85. Recording on an IC card from a magneto-optical disk or an optical disk by changing the bit rate mode and decreasing the bit rate is suitable for recording on an IC card having a high price per recording capacity. This is convenient because the sampling frequency is not the same regardless of the bit rate mode and unnecessary sampling frequency conversion is not required. The actual additional compression is performed by the additional compressor 84.

Next, a so-called high-speed digital dubbing operation will be described. First, at the time of so-called high-speed digital dubbing, the system controller 57 executes a predetermined high-speed dubbing control processing operation by operating a dubbing operation key or the like of the key input operation unit 58. Specifically, the compressed data from the decoder 71 is directly applied to the IC.
The data is sent to the memory 85 of the card recording system, subjected to variable bit rate coding by the additional compressor 84 that performs entropy coding, and recorded in the IC card 2 via the IC card interface circuit 86. Here, for example, when the stereo mode ATC data of the B mode is recorded on the magneto-optical disk 1, eight times compressed data is continuously read from the decoder 71.

Therefore, at the time of high-speed dubbing, compressed data corresponding to 8 times (in the case of the B-mode stereo mode) in real time is continuously obtained from the magneto-optical disk 1, which is entropy as it is. Since it is encoded and converted into a constant bit rate with a low bit rate and recorded in the IC card 2, 8 times high-speed dubbing can be realized. If the compression mode is different, the magnification of the dubbing speed is also different. Also, dubbing may be performed at a high speed equal to or higher than the compression ratio. In this case, the magneto-optical disk 1 is driven to rotate at high speed at a speed several times the steady speed.

By the way, as shown in FIG. 2, the bit-compressed data is recorded on the magneto-optical disk 1 at a constant bit rate, and at the same time, the data is subjected to variable bit compression by the additional compression / expansion block 3. Information of the amount of data when encoded (that is, the data recording capacity required for recording in the IC card 2) is recorded. By doing so, for example, the number of songs that can be recorded in the IC card 2 among the songs recorded on the magneto-optical disc 1, the combination of the songs, and the like can be immediately known by reading the data amount information. . Of course, additional compression / decompression block 84 may perform additional compression operations to lower bitrate modes of fixed bitrate rather than variable bitrate mode.

On the contrary, in the IC card 2, not only the data bit-compressed and encoded at the variable bit rate,
By recording the data amount information of the data bit-compressed and encoded at a constant bit rate, it is possible to quickly know the data amount when the data such as a song is sent from the IC card 2 to the magneto-optical disk 1 for recording. it can. Of course, not only the data bit-compressed and encoded at the variable bit rate but also the data bit-compressed and encoded at the constant bit rate can be recorded in the IC card 2.

FIG. 3 shows the external appearance of the compressed data recording and / or reproducing apparatus 5 having the structure shown in FIG. 1, in which a magneto-optical disk or optical disk insertion portion 6 and an IC card insertion slot 7 are provided. Has been. Of course, the disc and the IC card may be separate sets and a signal may be transmitted between them with a cable.

Next, the high efficiency compression coding will be described in detail. That is, refer to FIG. 4 and subsequent figures for a technique for highly efficient encoding of an input digital signal such as an audio PCM signal using band division encoding (SBC), adaptive transform encoding (ATC) and adaptive bit allocation techniques. While explaining.

In the specific high-efficiency coding apparatus shown in FIG. 4, first, the input digital signal is divided into a plurality of frequency bands, and the two adjacent lowest bands have the same bandwidth, and the higher frequency band. Then, selecting a wider bandwidth for higher frequency bands, performing orthogonal transformation for each frequency band,
The obtained spectrum data on the frequency axis is subdivided into the critical bandwidths in the low frequency range, taking into account the human auditory characteristics described later, in so-called critical bandwidths, and in the high and middle frequency range, the block floating efficiency is considered. Bits are adaptively allocated and encoded for each band. Normally, this block is the quantization noise generation block. further,
In the embodiment of the present invention, the block size (block length) is adaptively changed according to the input signal before the orthogonal transformation, and the floating process is performed for each block.

That is, in FIG. 4, the input terminal 100
For example, when the sampling frequency is 44.1 kHz,
An audio PCM signal of 0 to 22 kHz is supplied. This input signal is divided into a band of 0 to 11 kHz and a band of 11 to 22 kHz (high band) by a band division filter 101 such as a so-called QMF filter, and 0 to 1 is divided.
Similarly, a signal in the 1 kHz band is divided into a 0 to 5.5 kHz band (low band) and a 5.5 to 11 kHz band (middle band) by a band splitting filter 102 such as a so-called QMF filter. The signals of each band from the band division filters 101 and 102 are sent to the orthogonal transform block size determination circuit 106, and the block size is determined for each band. Here, in the orthogonal transform block size determination circuit 106, the length of the block size is basically 11.6 ms, which is the maximum block size. If the signal is quasi-stationary in time, set the orthogonal transform block size to 1
The maximum choice is 1.6 ms to increase the frequency resolution and 1 if the signal is non-stationary in time.
In the band of 1 kHz or less, the orthogonal transform block size is further divided into four, and in the band of 11 kHz or more, the orthogonal transform block size is divided into eight, thereby improving the time resolution.

As a method of dividing the above-mentioned input digital signal into a plurality of frequency bands, there is, for example, a QMF filter, and 1976 RECrochiere Digital coding.
of speech in subbands Bell Syst.Tech. J. Vol.55, No.
8 1976. Also ICASSP 83, BOSTON Pol
yphase Quadrature filters-A new subbandcodingtechn
ique Joseph H. Rothweiler describes a filter partitioning technique with equal bandwidth.

Referring again to FIG. 4, the band division filter 1
The outputs of 01 and 102 are supplied to the orthogonal transform circuits 103, 104 and 105 for each signal in each band. At the same time, the block size determined by the orthogonal transform size determining circuit 106 is the orthogonal transform circuits 103, 104, 1
05, and the filter output is divided into blocks according to the block size and subjected to orthogonal transform processing. Figure 5
Indicates the orthogonal transform block size, which is 11.6 ms (long mode) or 2.9 ms in the low and middle frequencies.
Select either (Short mode), 1 in high range
Select either 1.6 ms (long mode) or 1.45 ms (short mode). The determined orthogonal transform block size information is taken out from the terminal 111 and sent to the decoding circuit.

Here, as the above-mentioned orthogonal transform, for example, the input audio signal is divided into blocks in a predetermined unit time (frame), and the fast Fourier transform (F
FT), Discrete Cosine Transform (DCT), Modified Discrete Cosine Transform (MDCT) and the like are used to transform the time axis into the frequency axis. IC for MDCT
ASSP 1987 Subband / Transform Coding Using Filter Ba
nk Designs BasedonTime Domain Aliasing Cancellatio
n JPPrincen ABBradley Univ. of SurreyRoyal Melb
Ourne Inst.of Tech.

The bit allocation calculating circuit 107 masks each divided band in consideration of the critical band and block floating in consideration of so-called masking effect based on the spectrum data divided in consideration of the critical band and block floating. The amount is calculated, and the number of allocated bits is calculated for each band based on the masking amount and the energy or peak value for each divided band in consideration of the critical band and block floating. In the adaptive bit allocation coding circuit 108, each spectrum data (or MD) is calculated according to the number of bits allocated for each band by the bit allocation calculation circuit 107.
(CT coefficient data) is requantized. The data encoded in this way is taken out via the output terminal 110.

Next, FIG. 6 shows the bit allocation calculation circuit 10 described above.
7 is a block circuit diagram showing a schematic configuration of one specific example of No. 7;
In FIG. 6, the input terminal 21 is supplied with spectrum data on the frequency axis from the orthogonal transform circuits 103, 104, 105.

The input data on the frequency axis is sent to the energy calculation circuit 22 for each band, and the energy of each divided band considering the masking amount, the critical band and the block floating is, for example, each energy in the band. It can be obtained by calculating the sum of amplitude values. Instead of the energy for each band, a peak value, an average value, etc. of the amplitude value may be used. As an output from the energy calculation circuit 22, for example, the spectrum of the sum total value of each band is shown as SB in FIG. However,
In FIG. 7, in order to simplify the illustration, the number of divided bands in consideration of the masking amount, the critical band, and the block floating is represented by 12 bands (B1 to B12).

Here, in order to take into account the influence of the spectrum SB on so-called masking, a convolution process is performed such that the spectrum SB is multiplied by a predetermined weighting function and added. Therefore, the output of the energy calculation circuit 22 for each band, that is, each value of the spectrum SB is sent to the convolution filter circuit 23. The convolution filter circuit 23 includes, for example, a plurality of delay elements for sequentially delaying input data, and a plurality of multipliers (for example, 25 corresponding to each band, for multiplying outputs from these delay elements by a filter coefficient (weighting function)). Number of multipliers), and a sum adder that sums the outputs of the multipliers. By this convolution processing, the sum total of the portion shown by the dotted line in FIG. 7 is obtained. The masking refers to a phenomenon in which a certain signal masks another signal to make it inaudible due to human auditory characteristics, and this masking effect includes a time axis masking by an audio signal on the time axis. There are an effect and a simultaneous masking effect by a signal on the frequency axis. Due to these masking effects, even if there is noise in the masked portion, this noise cannot be heard. Therefore, in the actual audio signal, the noise within the masked range is regarded as an acceptable noise.

Here, a specific example of the multiplication coefficient (filter coefficient) of each multiplier of the convolution filter circuit 23 will be described. When the coefficient of the multiplier M corresponding to an arbitrary band is 1, the multiplier is M-1 gives a coefficient of 0.15, multiplier M-2 gives a coefficient of 0.0019, and multiplier M-3 gives a coefficient of 0.0000.
086, multiplier M + 1 gives a coefficient of 0.4, multiplier M + 2
By multiplying the output of each delay element by a coefficient of 0.06 with a coefficient of 0.007 with a multiplier M + 3, the convolution processing of the spectrum SB is performed. However, M is an arbitrary integer of 1 to 25.

Next, the output of the convolution filter circuit 23 is sent to the subtractor 24. The subtractor 24 obtains a level α corresponding to an allowable noise level described later in the convoluted area. The level α corresponding to the permissible noise level (permissible noise level)
Is a level at which an allowable noise level is obtained for each band of the critical band by performing the inverse convolution process as described later. Here, the subtractor 24 is supplied with an allowance function (function expressing a masking level) for obtaining the level α. The level α is controlled by increasing or decreasing this allowance function. The permissible function is (n-
ai) It is supplied from the function generating circuit 25.

That is, the level α corresponding to the allowable noise level can be obtained by the following equation (1), where i is the number given in order from the low band of the critical band. α = S- (n-ai) (1) In this equation (1), n and a are constants, a> 0, S is the intensity of the convolution-processed Bark spectrum, and (1)
In the formula, (n-ai) is the allowable function. In this embodiment, n =
38, a = 1, and there was no sound quality deterioration at this time, and good encoding was performed.

In this way, the level α is obtained, and this data is transmitted to the divider 26. The divider 26 is for inverse convolution of the level α in the convolved area. Therefore, by performing this inverse convolution process,
The masking spectrum can be obtained from the level α. That is, this masking spectrum becomes the allowable noise spectrum. Although the above-mentioned inverse convolution processing requires a complicated operation, in the present embodiment, the inverse convolution is performed using the simplified divider 26.

Next, the masking spectrum is transmitted to the subtractor 28 via the synthesizing circuit 27. Here, the subtractor 28 includes an energy calculation circuit 22 for each band.
From the output signal, that is, the spectrum SB described above is supplied via the delay circuit 29. Therefore, the masking spectrum and the spectrum SB are obtained by the subtractor 28.
By performing the subtraction operation with and, the spectrum SB is masked below the level indicated by the level of the masking spectrum MS, as shown in FIG.

The output from the subtracter 28 is taken out via the allowable noise correction circuit 30 and the output terminal 31,
For example, the allocation bit number information is sent to a ROM or the like (not shown) in which it is stored in advance. This ROM or the like is assigned to each band in accordance with the output (the level of the difference between the energy of each band and the output of the noise level setting means) obtained from the subtraction circuit 28 via the allowable noise correction circuit 30. Outputs bit number information. By transmitting this allocation bit number information to the adaptive bit allocation encoding circuit 108, each spectrum data on the frequency axis from the orthogonal transform circuits 103, 104, 105 is quantized by the number of bits allocated for each band. Will be converted.

In summary, in the adaptive bit allocation coding circuit 108, according to the level of the difference between the energy of each divided band considering the masking amount, the critical band and the block floating, and the output of the noise level setting means. The spectrum data for each band is quantized by the allocated number of bits. In addition,
The delay circuit 29 considers the amount of delay in each circuit before the synthesis circuit 27 and the spectrum S from the energy detection circuit 22.
It is provided to delay B.

By the way, at the time of synthesizing by the above-mentioned synthesizing circuit 27, data showing a so-called minimum audible curve RC which is the human auditory characteristic as shown in FIG. The masking spectrum MS can be combined. In this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Even if the coding is the same, the minimum audible curve is different depending on, for example, a difference in reproduction volume at the time of reproduction, but in a realistic digital system, for example, the way music enters the 16-bit dynamic range is very different. Therefore, if the quantization noise in the most audible frequency band around 4 kHz is not heard, it is considered that the quantization noise below the level of the minimum audible curve is not heard in other frequency bands. Therefore, it is assumed that the system is used in such a manner that the noise near the 4 kHz of the word length of the system cannot be heard, and the minimum audible curve RC
When the allowable noise level is obtained by synthesizing the masking spectrum MS and the masking spectrum MS together, the allowable noise level in this case can be up to the shaded portion in FIG. 9. In this embodiment, the level of 4 kHz of the minimum audible curve is set to the minimum level equivalent to 20 bits, for example. Further, FIG. 9 also shows the signal spectrum SS at the same time.

The allowable noise correction circuit 30 corrects the allowable noise level in the output from the subtractor 28 based on the information of the equal loudness curve sent from the correction information output circuit 33. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics, for example, a curve obtained by obtaining the sound pressure of sound at each frequency that sounds the same as a pure tone of 1 kHz, and connecting the curves. Also called sensitivity curve. Further, this equal loudness curve draws a curve substantially the same as the minimum audible curve RC shown in FIG. In this equal loudness curve, for example, in the vicinity of 4 kHz, even if the sound pressure drops by 8 to 10 dB from 1 kHz, it sounds as loud as 1 kHz. It doesn't sound the same. Therefore, it is understood that it is preferable that the noise (allowable noise level) exceeding the level of the minimum audible curve has a frequency characteristic given by a curve corresponding to the equal loudness curve. From this, it can be seen that correcting the permissible noise level in consideration of the equal loudness curve is suitable for human hearing characteristics.

Here, the correction information output circuit 33 is provided between the detection output of the output information amount (data amount) at the time of quantization in the encoding circuit 108 and the bit rate target value of the final encoded data. The allowable noise level may be corrected based on the error information. This is because the total number of bits obtained by performing temporary adaptive bit allocation in advance for all bit allocation unit blocks is a fixed number of bits (target value) determined by the bit rate of the final encoded output data. May have an error with respect to, and bit allocation is performed again so that the error may be zero. That is, when the total allocated bit number is smaller than the target value, the difference bit number is allocated to each unit block and added, and when the total allocated bit number is larger than the target value, the difference bit number is added to each unit block. It is allotted to and scraped.

In order to do this, an error in the total allocation bit number from the target value is detected, and the correction information output circuit 33 corrects the allocation bit number according to the error data. Is output. Here, when the error data indicates a bit number shortage, it can be considered that the data amount is larger than the target value because a large number of bits are used per unit block. Further, when the error data is data indicating a surplus of the number of bits, it is possible to consider a case where the number of bits per unit block is small and the amount of data is smaller than the target value. Therefore, the correction information output circuit 33 outputs the correction value for correcting the allowable noise level in the output from the subtracter 28 based on the error data, for example, based on the information data of the equal loudness curve. Data will be output. By transmitting the correction value as described above to the allowable noise correction circuit 30, the allowable noise level from the subtractor 28 is corrected. In the system as described above, the data obtained by processing the orthogonal transform output spectrum by the sub information as the main information, the scale factor indicating the block floating state and the word length indicating the word length are obtained as the sub information, and the data is sent from the encoder to the decoder. To be

Here, the bit allocation calculation circuit 107
Can also be configured as shown in FIG. This Figure 1
By using 0, the following effective bit allocation method different from the bit allocation method described above will be described.

Each MDCT circuit 103 in FIG.
The outputs of 104 and 105 are energy calculation circuit 3 for calculating the energy for each band via the input terminal 300 of FIG.
Sent to 01. Energy calculation circuit 301 for each band
Then, the energy of each band obtained by further dividing the critical band in the critical band or the high band is obtained by, for example, calculating the square root of the root mean square of each amplitude value in the band. Instead of the energy for each band, it is possible to use the peak value or average value of the amplitude values.

As the output from the energy calculation circuit 301, for example, the spectrum of the sum value for each critical band (critical band) or for each band obtained by further dividing the critical band in the high frequency band is, for example, the spectrum shown in FIG. Burk spectrum) SB.

Here, in this embodiment, the number of bits that can be used for transmission or recording by expressing the MDCT coefficient is, for example, 1 k.
If it is a bit / block, one of them is used in this embodiment.
Create a fixed bit allocation pattern using k bits.
In this embodiment, a plurality of bit allocation patterns for the fixed bit allocation are prepared, and various selections can be made depending on the characteristics of the signal. In the present embodiment, the fixed bit allocation circuit 307 has various patterns in which the bit amount of the short-time block corresponding to 1 k bits is distributed to each frequency. In the fixed bit allocation circuit 307, in particular, a plurality of patterns having different bit allocation ratios for the middle low band and the high band are prepared. Then, as the signal size is smaller, a pattern with a smaller amount of allocation to the high frequency band is selected.
By doing so, the loudness effect in which the sensitivity in the high frequency range decreases as the signal becomes smaller can be used. As the signal magnitude at this time, the magnitude of the signal in the entire band may be used. For example, the output of the non-blocking frequency division circuit using a filter or the orthogonal transform output such as the MDCT output may be used. It can also be used. The 1 k bit (the number of usable bits) is set by the total usable bit number input circuit 305, for example.
This total number of usable bits can be input from the outside.

The output from the energy calculation circuit 301 is also sent to the energy-dependent bit allocation circuit 306. The energy-dependent bit allocation circuit 306 determines an energy-dependent bit allocation pattern from the energy of each band. The energy-dependent bit pattern based on this energy is allocated such that more bits are allocated as the energy of the band is larger.

In FIG. 10, the division ratio between the above-mentioned allocation to the fixed bit allocation pattern and the bit allocation depending on the Bark spectrum (spectrum SB) is determined by the index (tonality) indicating the smoothness of the signal spectrum. To be done. That is, in this embodiment, the output of the energy calculation circuit 301 is converted into the spectrum smoothness calculation circuit 3
02, the spectrum smoothness calculation circuit 302 calculates a value obtained by dividing the sum of absolute values of differences between adjacent values of the signal spectrum by the sum of the signal spectrum, and uses this value as an index (tonality). . When the tonality is determined, the bit division rate determination circuit 304 determines the division rate. The division ratio is a value for changing the weighting of fixed bit allocation and energy-dependent bit allocation.

The division rate data from the bit division rate determining circuit 304 is supplied with a multiplier 312 to which the output of the fixed bit allocation circuit 307 is supplied and a multiplier 311 to which the output of the energy dependent bit allocation circuit 306 is supplied. Sent to. The outputs of these multipliers 312 and 311 are sent to the sum calculation circuit 308. That is, the fixed bit allocation and the bit allocation value depending on the energy of each band (critical band or band in which the critical band is further subdivided in the high band) are multiplied by the above-mentioned division ratio, respectively, and these two values are obtained. Are added in the sum calculation circuit 308, and the result of this operation is output terminal (bit allocation amount output terminal of each band) 309.
To the structure of the latter stage and used for quantization and encoding.

The state of bit allocation at this time is shown in FIG.
(b) and (b) of FIG. The state of quantization noise corresponding to this is shown in (a) of FIG. 11 and (a) of FIG. 11 (a) and 11 (b) show the case where the signal spectrum is relatively flat, and FIGS. 12 (a) and 12 (b) show the case where the signal spectrum shows a high tonality. Also,
In FIGS. 11 (b) and 12 (b), Q _S represents the signal level-dependent bit amount, and Q _F in the diagrams represents the fixed bit allocation bit amount. 11 (a) and FIG. 12
figure L in (a) shows the signal level, noise reduction caused by N _S is the signal level dependent component in the figure, reference numeral N _F indicates the noise level due to fixed bit allocation amount.

If the spectrum of the signal is relatively flat
In FIG. 11 showing the combination, a large number of fixed bits are usually used.
Bit allocation based on the total allocation
And helps to obtain a large signal-to-noise ratio. But this
In the case of Fig. 11 of Fig. 11, it is relatively small in the low range and high range.
Bit allocation will be used. This is hearing
This is because the importance of this band is small. Also this
When Q in FIG. _SAs shown in, some signal level
Depending on the number of bits (bits) that
The noise level in the band with a large signal is selectively reduced.
Sent. Therefore, the spectrum of the signal is fairly flat.
In some cases, this selectivity also works over a relatively wide bandwidth
It will be.

On the other hand, as shown in FIG. 12, when the signal spectrum exhibits high tonality, a large amount of signal level-dependent bit allocation is performed as shown by Q _S in the drawing of FIG. The reduction of the quantization noise by) is used to reduce the noise in a very narrow band (band indicated by N _S in the drawing of FIG. 12). This achieves improved performance with isolated spectrum input signals. In addition, the noise level in a wide band is non-selectively lowered by the bit allocation that is performed by a bit of fixed bit allocation at the same time.

Again, referring to FIG. 4, the adaptive bit allocation encoding circuit 108 will be described. In the present embodiment, for example, there are two kinds of bit rate modes, for example, A
The mode is 128 kbps / chnnel, and the B mode is 64 kbps / chnnel, which is half the A mode. In addition, the present embodiment is not limited to two types of modes, and it is possible to have a plurality of modes.

First, the encoding method in the A mode will be described. 13 and 14 show a specific example of block floating band division in the A mode.
FIG. 13 shows the case where the orthogonal transform block size is 11.6 ms, and FIG. 14 shows the case where the orthogonal transform block size is 4 in the low and middle range.
In the case of division and high frequency division, there are eight divisions, but in both cases, the total number of block floating bands is the same, and division is made into 52 bands. Further, looking at each band that is the output of the band division filter, there are 20 block floating in the low band and 16 block floating in the middle and high band, and this number is determined regardless of the orthogonal transform block size. There is no problem if the block size changes independently for each band. For example, in the case where the block size is 11.6 ms divided into four only in the low frequency range and the block size in the middle and high frequencies is 11.6 ms, the block floating band is shown in FIG.
If divided as 3, the total number of bands is 52. In the adaptive bit allocation encoding circuit 108, this 52
The scale factor and word length information is given to each block floating band, and the spectrum data is quantized and coded according to the given scale factor word length.

The encoded data is taken out from the terminal 110 and recorded or transmitted.

Next, the B-mode coding method will be described. Since the bit rate in B mode is half that in A mode, the amount of sub information (scale factor, word length, etc.) does not change and the amount of main information (spectral data) does not change when encoded in the same way as in A mode. As compared to the A mode, the proportion of sub information in the total amount of information increases and the coding efficiency decreases. When the bit rate is halved, it is desirable to reduce not only the main information amount but also the sub information amount by half or less. In the present embodiment, in order to reduce the amount of sub information in B mode to half that in A mode, the value of sub information is shared between two blocks that are temporally adjacent to each other, thereby reducing the amount of sub information. is doing.
That is, since the sub information amount in the A mode is basically equal to the number of block floating bands, 52 pieces / 1
Although it is 1.6 ms, in the B mode, since the time axis direction of the block floating band is extended, it becomes 52 pieces / 23.2 ms, and comparing the sub information amounts in the same time, it is compared with the A mode. It is half the amount. 15, 16 and 17 show specific examples of block floating band division in the B mode.

FIG. 15 shows a case where the orthogonal transform block sizes of two blocks temporally adjacent to each other are both in the long mode. The region surrounded by the solid line is the orthogonal transform block and the region shaded is one. Represents one block floating band. That is, this block floating band is a combination of two adjacent bands in the time axis direction of the A mode block floating band in FIG. 13, and the band division in the frequency axis direction is exactly the same as in FIG.

FIG. 16 shows a case where the orthogonal transform block sizes of two blocks temporally adjacent to each other are both in the short mode. As in FIG. 15, the region surrounded by a solid line is an orthogonal transform block and is displayed in diagonal lines. The region shown in the figure represents one block floating band. That is, this block floating band combines two adjacent bands in the time axis direction of the A mode block floating band in FIG. 14, and the band division in the frequency axis direction is exactly the same as in FIG.

FIG. 17 shows a case where two blocks which are temporally adjacent to each other have different orthogonal transform block sizes, that is, a combination of a short mode and a long mode.
Similarly, a region surrounded by a solid line represents an orthogonal transformation block, and a region shaded by a diagonal line represents one block floating band. Blocks whose orthogonal transform block size is the short mode (0 to 11.6 in FIG. 17)
The middle range of ms and the low range and high range of 11.6 to 23.2 ms are the same as those in the short mode (FIG. 16), that is, the time of the block floating band in A mode in FIG. Two bands adjacent to each other in the axial direction are combined into one band, and the band division in the frequency axial direction is exactly the same as that in FIG. Conversely, a block in which the orthogonal transform block size is the long mode (a low band of 0 to 11.6 ms and a high band of 11.6 to 23.2 m in FIG. 17).
In the middle region of s), the bands cannot be put together in the time axis direction, so that two bands adjacent to each other in the frequency axis direction are exceptionally put together into one band. Exactly the same.

As described above, in order to reduce the number of sub-information in the B mode to half that in the A mode, the block floating bands adjacent to each other in the time axis direction or the frequency axis direction are shared, resulting in bit The coding efficiency is improved by preventing an extreme decrease in main information due to the rate decrease.

FIG. 18 shows a specific example of the adaptive bit allocation coding circuit 108 in the B mode, in which the terminal 401 has orthogonal transform block size information, and the terminal 402.
Are given spectral data (MDCT coefficients) for the two blocks. The orthogonal transform block size information is stored in the scale factor reset circuit 405,
The spectrum data (MDCT coefficient) is sent to the word length reset circuit 406, and sent to the quantizer 408.

Further, the scale factor A supplied via the input circuit 403 and set for each band in the block floating band division for the A mode is shared by the scale factor resetting circuit 405 as described above. The values of the two block floating bands to be combined are combined and the scale factor B for B mode is reset. Usually, the larger one of the two scale factors A is selected and set as a common scale factor.

Similarly, the word length A set for each band by the block floating band division for the A mode supplied through the input circuit 404 is the word length for the B mode in the word length reset circuit 406. B is reset. When the word lengths are made common, for example, the larger one of the two word lengths A is selected. Alternatively, the average value of the two word lengths A or the like may be used.

The scale factor A and the word length A are sent to the resetting circuits 405 and 406 by using information of two blocks (23.3 ms) as one unit.

Next, the word length reset circuit 406.
In, the reset word length is corrected in the total bit number correction circuit 407 for the error in the total bit number caused by the reset. The reset scale factor B and word length B are both sent to the quantizer 408 and the encoder 409, and are used when quantizing the spectrum data. The quantized and encoded spectrum data is taken out as the encoded data B from the terminal 410.

Up to this point, the function of the coding apparatus for coding the time-series PCM signal has been described. Next, when the A-mode coded data is converted to the B-mode coded data, and when the B-mode coded A case of converting encoded data into encoded data of A mode will be described with reference to a specific example of FIG.

First, when converting from the A mode to the B mode, in FIG. 19, the input terminal 501 is supplied with the encoded data A encoded in the A mode, and the input terminal 503 is encoded. The orthogonal transform block size information is given. The orthogonal transform block size information is converted in the code converter 508 from the code representing the orthogonal transform block size in A mode to that in B mode, sent to the bit allocation calculation circuit 507, and taken out from the output terminal 513.

The function of the code converter 508 is simply to express the two blocks of coded A mode orthogonal transform block size information in a B mode code, and change the meanings of both. There is no.

The coded data A is sent to the A-mode adaptive bit allocation decoding circuit 505, where it is decoded and dequantized to be restored to spectrum data. The obtained spectrum data is sent to the bit allocation calculation circuit 507, and bits are allocated. This bit allocation calculation circuit 507
Has the same function as the bit allocation calculation circuit 107 described above.

Here, the restored spectrum data is B
It is sent to the mode adaptive bit allocation encoding circuit 506, and the above-mentioned B mode encoding is performed. The quantized and encoded coded data B is taken out from the output terminal 511. As described above, the conversion from the A mode to the B mode can be performed by a simple circuit in which the A mode decoding circuit and the B mode encoding circuit are combined, and high speed conversion is possible.

Next, when converting from B mode to A mode, similarly, in FIG. 19, the input terminal 512 is supplied with the coded data B coded in B mode,
Coded orthogonal transform block size information is given to the input terminal 514. The orthogonal transform block size information is converted in the code converter 509 from the code representing the orthogonal transform block size in B mode to that in A mode, sent to the format conversion circuit 510, and taken out from the output terminal 504.

The function of the code converter 509 is exactly the opposite of that of the code converter 508, and the coded B-mode orthogonal transform block size information is divided into codes for two blocks for A-mode. Just do.

The coded data B is sent to the format conversion circuit 510, and the coded data as it is is directly converted into the format of the A mode and taken out from the output terminal 502. In this case, there is substantially no change in bit rate between the two modes, and the main information is used only about half in the A mode format. Also,
Although it is possible to use a method of decoding, re-allocating the bits again, and then encoding as in the conversion from the A mode to the B mode, even if the substantial amount of information increases, the sound quality deteriorates due to requantization. To do. As described above, the conversion from the B mode to the A mode is performed only by the format conversion, that is, the processing of only the simple rearrangement of the encoded codes, so that the conversion at a higher speed is possible.

Next, the decoding device will be described. Figure 20
4, the input terminal 210 is supplied with the coded data on the frequency axis obtained from the output terminal 110 of FIG. 4, and this coded data is first sent to the decoding circuit 208 for adaptive bit allocation to perform the decoding process. Then, the spectrum data on the frequency axis is restored.

The orthogonal transform block size information from the above coding device is given to the input terminal 211 and is supplied to the inverse orthogonal transform circuits 203, 204 and 205 for each band. Here, 0 to 5.5 of the above spectrum data
The data in the kHz band is sent to the inverse orthogonal transform circuit 203 by 5.5.
The data in the k to 11 kHz band is the inverse orthogonal transform circuit 204.
The data in the 11 kHz to 22 kHz band is sent to the inverse orthogonal transform circuit 205, and these circuits perform inverse orthogonal transform processing for each band in accordance with the orthogonal transform block size information.

Furthermore, the inverse orthogonal transform circuits 204 and 20 are also provided.
The output of No. 5 is combined by the band synthesizing filter 202, and the outputs of the inverse orthogonal transform circuit 203 and the synthesizing filter 202 are combined by the synthesizing filter 201 to be a reproduction signal, which is taken out from the output terminal 200.

The present invention is not limited to the above-described embodiment. For example, the one recording / reproducing medium and the other recording / reproducing medium do not need to be integrated, and data is transferred between them. It is also possible to tie it with a cable. Furthermore, for example, the present invention can be applied not only to audio PCM signals but also to signal processing devices for digital audio (speech) signals, digital video signals, and the like. Further, the above-described minimum audible curve synthesizing process may not be performed. In this case, the minimum audible curve generating circuit 32 and the synthesizing circuit 27 are unnecessary, and the output from the subtractor 24 is the divider 26.
After being inversely convolved with, the data is immediately transmitted to the subtractor 28. There are various types of bit allocation methods, and the simplest is to use fixed bit allocation, simple bit allocation by each band energy of a signal, or bit allocation combining fixed and variable components.

[0104]

As is apparent from the above description, according to the compressed data recording and / or reproducing apparatus, method or recording medium according to the present invention, the same regardless of the difference in bit rate between plural modes. By using the sampling frequency, it is possible to prevent the sampling frequency signal generation circuit from becoming complicated and the hardware scale from increasing when a plurality of sampling frequencies are provided.

In the case of encoding at a low bit rate, so-called sub information is shared by a plurality of blocks adjacent in the time axis direction or a plurality of so-called block floating bands in the frequency axis direction within the same time block. By recording or transmitting, the amount of sub information can be reduced, and by assigning the reduced amount of sub information to the main information, the coding efficiency can be improved.

Further, when it is desired to convert the compressed signal in the high bit rate mode to the low bit rate mode for recording for a longer period of time, it is not necessary to convert the original compressed signal from the frequency axis to the time axis. Since a low bit rate compressed signal can be obtained only by data conversion on the frequency axis and the processing steps of orthogonal transformation, inverse orthogonal transformation and band division / synthesis filter can be omitted, high-speed signal conversion can be performed. I can.

[Brief description of drawings]

FIG. 1 is a block circuit diagram showing a configuration example of a recording / reproducing apparatus as an embodiment of a compressed data recording / reproducing apparatus according to the present invention.

FIG. 2 is a diagram showing recorded contents of a magneto-optical disk 1 and an IC card.

FIG. 3 is a schematic front view showing an example of the external appearance of the device of this embodiment.

FIG. 4 is a block circuit diagram showing a specific example of an encoding device that realizes the audio high efficiency encoding method according to the present embodiment.

FIG. 5 is a diagram for explaining an orthogonal transform block size according to the present embodiment.

FIG. 6 is a block circuit diagram showing a specific configuration of a bit allocation calculation function.

FIG. 7 is a diagram showing spectra of bands divided in consideration of each critical band and block floating.

FIG. 8 is a diagram showing a masking spectrum.

FIG. 9 is a diagram in which a minimum audible curve and a masking spectrum are combined.

FIG. 10 is a block circuit diagram showing a specific configuration for realizing the second bit allocation method.

FIG. 11 is a diagram showing a noise spectrum and a bit allocation when the signal spectrum is flat in the second bit allocation method.

FIG. 12 is a diagram showing a noise spectrum and bit allocation when the tonality of a signal spectrum is high in the second bit allocation method.

FIG. 13 is a diagram regarding frequency and time indicating a frequency band of 52 divisions in consideration of a critical band and a block floating in a processing block of 11.6 ms in A mode (orthogonal transform block size is long mode).

FIG. 14 is a diagram regarding frequency and time showing a frequency band of 52 divisions in consideration of a critical band and a block floating in a processing block of 11.6 ms in A mode (orthogonal transform block size is short mode).

FIG. 15 is a diagram regarding frequency and time showing a frequency band of 52 divisions in consideration of a critical band and a block floating in a processing block of 23.2 ms in the B mode (both orthogonal transform block sizes are long modes).

FIG. 16 is a diagram regarding frequency and time showing a frequency band of 52 divisions in consideration of a critical band and a block floating in a processing block of 23.2 ms in B mode (both orthogonal transform block sizes are short mode).

FIG. 17 is a diagram regarding frequency and time indicating a frequency band of 52 divisions in consideration of a critical band and a block floating in a processing block of 23.2 ms in B mode (combination of long mode and short mode for orthogonal transform block size). ).

FIG. 18 is a block diagram showing a specific example of an adaptive bit allocation encoding circuit in B mode.

FIG. 19 is a block circuit diagram showing a specific configuration for performing high-speed conversion from A mode to B mode.

FIG. 20 is a block circuit diagram showing a specific example of a decoding device that realizes the high-efficiency audio encoding method according to the present embodiment.

[Explanation of symbols]

1 ・ ・ Optical disk 2 ・ ・ ・ ・ IC card 3 ・ ・ ・ ・ Additional compression / expansion function block 5 ・ ・ ・ ・ Recording / playback device 6 ・ ・ ・ ・ ・ ・ Optical disk slot 7 ... IC card slot 53 Optical head 54 Magnetic head 56 Servo control circuit 57 System controller 62, 83 A / D converter 63 ... ATC encoder 64, 72, 85 ... Memory 65 ... Encoder 66 ... Magnetic head drive circuit 71 ... Decoder 73 ... .. ATC decoder 74 .. D / A converter 100 .. .. Acoustic signal input terminals 101, 102 .. Band division filter 103 .. .. High frequency orthogonal transform circuit (MDCT) 104 ..・ Mid-range orthogonal transform circuit (M CT) 105 ... Low-frequency orthogonal transform circuit (MDCT) 106 ... Orthogonal transform block size determination circuit 107 ... Bit allocation calculation circuit 108 ... Adaptive bit allocation coding circuit 110 ... -Encoding output terminal 111 --- Orthogonal transform block size information output terminal 21-Allowable noise level calculation function input terminal 22-Energy detection circuit for each band 23-Convolution Filter circuit 24 ... Subtractor 25 ... (n-ai) function generating circuit 26 ... Divider 27 ... Combining circuit 28 ... Subtractor 30 ... Permissible noise correction circuit 31 ... Permissible noise level calculation function output terminal 32 ... Minimum audible curve generation circuit 33 ... Correction information output circuit 300 ... Transform output (MDCT coefficient Input terminal 301 ... Energy calculation circuit for each band 302 ... Spectrum smoothness calculation circuit 304 ... Bit division ratio determination circuit 305 ... Total usable bit number input circuit 306 ... Energy-dependent bit allocation circuit 307 ... Fixed bit allocation circuit 308 ... Bit sum operation circuit 309 ... Bit allocation amount output terminal for each band 401 ... Orthogonal transform block size Information input terminal 402 ... Spectral data input terminal 403 ... Input circuit of scale factor A set in A mode 404 ... Word length A set in A mode
Input circuit 405 ...- Scale factor resetting circuit 406 ...- Word length resetting circuit 407 ...- Total bit number correction circuit 408 ... Quantizer 409 ... Encoding 410 ... Coded data output terminal 501 ... A mode coded data input terminal 502 ... A mode coded data output terminal 503 ... A mode orthogonal transform block size information input Terminal 504 ... A mode orthogonal transform block size information output terminal 505 ... A mode adaptive bit allocation decoding circuit 506 ... B mode adaptive bit allocation encoding circuit 507 ... Bit allocation Calculation circuit 508 ... Code converter 509 ... Code converter 510 ... Format conversion circuit 511 ... Output of B mode encoded data Terminal 512 ... B-mode encoded data input terminal 513 ... Orthogonal transform block size output terminal 514 ... Orthogonal transform block size input terminal 200 ... Acoustic signal output terminal 201, 202 ... -Band synthesis filter 203 --- High-frequency inverse orthogonal transform circuit 204 --- Middle-range inverse orthogonal transform circuit 205 --- Low-range inverse orthogonal transform circuit 208 --- Adaptive bit allocation decoding circuit 210- ... Encoded data input terminal 211 ... Orthogonal transform block size information input terminal

 ─────────────────────────────────────────────────── ─── Continuation of the front page (72) Inventor Makoto Mitsuno 6-735 Kita-Shinagawa, Shinagawa-ku, Tokyo Sony Corporation

Claims (61)

[Claims] 1. A digital signal is decomposed into a plurality of frequency band components to obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, and the information is compressed by quantizing each two-dimensional block regarding the time and frequency. , Recording or transmitting together with the information compression parameter for each two-dimensional block relating to the time and frequency, and / or expressing the signal component in the plurality of two-dimensional blocks relating to the time and frequency for which information compression has been performed, In a compressed data recording and / or reproducing or transmitting and / or receiving device which reproduces or receives using information compression parameters for each block, information compressing parameters of at least two two-dimensional blocks are collectively recorded or transmitted, and / or , Recording or reproducing compressed data characterized by reproducing, receiving It is properly transmitted and / or receiving device. 2. The information compression parameters of at least two two-dimensional blocks arranged at least in the time direction are summarized,
Recording and / or transmission, and / or reproduction or reception, compressed data recording and / or recording according to claim 1.
Or a reproducing or transmitting and / or receiving device. 3. The compressed data recording according to claim 1, wherein information compression parameters of at least two two-dimensional blocks arranged at least in the frequency direction are collectively recorded or transmitted and / or reproduced or received. / Or playback or transmission and / or receiving device. 4. The information compression parameters of at least two two-dimensional blocks arranged in the time direction and / or the frequency direction are collectively recorded or transmitted, and / or reproduced or received. The described compressed data recording and / or reproducing or transmitting and / or receiving device. 5. The compressed data according to claim 2, wherein the information compression parameters of at least two adjacent two-dimensional blocks arranged at least in the time direction are collectively recorded or transmitted and / or reproduced or received. Recording and / or reproducing or transmitting and / or receiving device. 6. The compressed data according to claim 3, wherein the information compression parameters of at least two adjacent two-dimensional blocks arranged at least in the frequency direction are collectively recorded or transmitted and / or reproduced or received. Recording and / or reproducing or transmitting and / or receiving device. 7. The information compression parameters of at least two adjacent two-dimensional blocks arranged in the time direction and / or the frequency direction are collectively recorded or transmitted, and / or reproduced or received. 6, the compressed data recording and / or reproducing or transmitting and / or receiving device. 8. A digital signal is decomposed into a plurality of frequency band components to obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, and the information is compressed by quantizing each two-dimensional block regarding the time and frequency. , Recording or transmitting together with the information compression parameter for each two-dimensional block relating to the time and frequency, and / or expressing the signal component in a plurality of two-dimensional blocks relating to the time and frequency where the information is compressed, two-dimensionally relating to the time and frequency. In a compressed data recording and / or reproducing or transmitting and / or receiving device which reproduces or receives using information compression parameters for each block, a mode for recording and / or reproducing a plurality of information bit rates or a mode for transmitting and / or receiving And the number of information compression parameters changes according to the information bit rate of each mode. Compressed data recording and / or reproducing or transmitting and / or receiving apparatus, characterized in Rukoto. 9. The compressed data recording and / or reproducing or transmitting according to claim 8, wherein a higher information bit rate mode records or transmits and / or reproduces or receives a larger number of information parameters. as well as/
Or a receiving device. 10. A mode having a lower information bit rate,
Record or transmit fewer information parameters, and / or
Alternatively, it is played back or received.
The described compressed data recording and / or reproducing or transmitting and / or receiving device. 11. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 9, wherein the number of information compression parameters changes in proportion to the information bit rate of each mode. 12. The information compression parameters of at least two two-dimensional blocks arranged in the time direction or the frequency direction or the time and frequency directions are collectively recorded or transmitted and / or reproduced or received, thereby The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 11, which achieves a reduction in the number. 13. Information compression by collectively recording or transmitting and / or reproducing or receiving information compression parameters of at least two adjacent two-dimensional blocks arranged in the time direction or the frequency direction or the time and frequency directions. Compressed data recording and / or reproducing or transmitting and / or receiving device according to claim 11, characterized in that a reduction in the number of parameters is achieved. 14. A mode having a lower information bit rate,
14. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 12, wherein a larger number of information compression parameters are collectively recorded or transmitted and / or reproduced or received. 15. The information compression parameter according to claim 14, wherein the number of information compression parameters to be grouped is made inversely proportional to the information bit rate for each mode, and a larger number of information compression parameters are grouped in a mode having a lower information bit rate. Compressed data recording and / or reproducing or transmitting and / or receiving device. 16. The compressed data recording and / or reproduction or transmission and / or reception according to claim 15, wherein the two-dimensional block is a block for block floating and / or a quantization noise generation control block. apparatus. 17. The input signal is an audio signal, and the frequency width of at least most of the quantization noise generation control blocks is made wider in a higher frequency band.
16. A compressed data recording and / or reproducing or transmitting and / or receiving device according to 16. 18. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 17, wherein the sampling frequencies of all modes are the same. 19. Using an orthogonal transform to decompose a digital signal into a plurality of frequency components to obtain a signal in a two-dimensional block of time and frequency, and / or in a two-dimensional block of time and frequency. 17. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 7, wherein inverse orthogonal transformation is used for converting the signal into a digital signal on the time axis. 20. In order to decompose a digital signal into a plurality of frequency band components and obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, first divide the digital signal into a plurality of bands, and for each of the divided bands. For each block of each band, a block consisting of multiple samples is formed, coefficient data is obtained by performing orthogonal transform for each block of each band, and / or conversion from multiple bands on the frequency axis to time axis signals is performed. 20. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 19, wherein inverse orthogonal transformation is performed and respective inverse orthogonal transformation outputs are combined to obtain a combined signal on a time axis. 21. The block size of orthogonal transform and / or inverse orthogonal transform is variable.
20. A compressed data recording and / or reproducing or transmitting and / or receiving device according to 20. 22. A division frequency width in division of a time axis signal before orthogonal transformation into a plurality of bands on a frequency axis and / or a combination of a plurality of bands on a frequency axis after inverse orthogonal transformation into a time axis signal Combined frequency width from multiple bands,
22. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 20, wherein the two consecutive lowest bands are the same. 23. A division frequency width in dividing a time axis signal before orthogonal transformation into a plurality of bands on a frequency axis and / or a combination of a plurality of bands on a frequency axis after inverse orthogonal transformation into a time axis signal Combined frequency width from multiple bands,
23. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 22, characterized in that it is widened in a substantially high region. 24. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 23, wherein a modified discrete cosine transform is used as the orthogonal transform and an inverse modified discrete cosine transform is used as the inverse orthogonal transform. . 25. Compressed data recording and / or reproducing or transmitting and / or receiving according to claim 23, wherein the transform block size of orthogonal transform is the same between all modes having different information bit rates. apparatus. 26. When converting compressed data between modes having different information bit rates, the compressed data is converted with the signal component on the frequency axis as it is without returning to the signal component on the time axis. 26. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 25. 27. When converting compressed data from a mode having a higher information bit rate to a mode having a lower information bit rate, the signal component on the frequency axis remains as it is without returning to the signal component on the time axis. 27. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 26, wherein the compressed data is converted by the method. 28. When converting compressed data from a mode having a lower information bit rate to a mode having a higher information bit rate, the encoded data is not decoded but is in an encoded state. 28. The compressed data recording and / or reproducing or transmitting and / or receiving apparatus according to claim 27, wherein the compressed data is converted as it is. 29. A digital signal is decomposed into a plurality of frequency band components to obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, and the information is compressed by quantizing each two-dimensional block regarding the time and frequency. , Recording or transmitting together with information compression parameters for each time-frequency two-dimensional block, and / or information-compressed signal components in a plurality of time-frequency two-dimensional blocks, the time-frequency two-dimensional block In the compressed data recording and / or reproducing or transmitting and / or receiving method of reproducing or receiving by using the information compressing parameter of each, the information compressing parameters of at least two two-dimensional blocks are collectively recorded or transmitted, and / or Compressed data recording and / or reproducing characterized by reproducing or receiving The transmission and / or reception method properly. 30. The compressed data recording according to claim 29, wherein the information compression parameters of at least two two-dimensional blocks arranged at least in the time direction are collectively recorded or transmitted and / or reproduced or received. And / or reproduction or transmission and / or reception method. 31. The compressed data recording method according to claim 29, wherein information compression parameters of at least two two-dimensional blocks arranged at least in the frequency direction are collectively recorded or transmitted and / or reproduced or received. And / or reproduction or transmission and / or reception method. 32. The information compression parameters of at least two two-dimensional blocks arranged in the time direction and / or the frequency direction are collectively recorded or transmitted, and / or reproduced or received. A method for recording and / or reproducing or transmitting and / or receiving the described compressed data. 33. The compressed data according to claim 30, wherein information compression parameters of at least two adjacent two-dimensional blocks arranged at least in the time direction are collectively recorded or transmitted and / or reproduced or received. Recording and / or reproduction or transmission and / or reception method. 34. The compressed data according to claim 31, wherein the information compression parameters of at least two adjacent two-dimensional blocks arranged at least in the frequency direction are collectively recorded or transmitted and / or reproduced or received. Recording and / or reproduction or transmission and / or reception method. 35. Information compression parameters of at least two adjacent two-dimensional blocks arranged in the time direction and / or the frequency direction are collectively recorded or transmitted, and / or
34. Reproducing or receiving the data.
4. The compressed data recording and / or reproducing or transmitting and / or receiving method described in 4. 36. A digital signal is decomposed into a plurality of frequency band components to obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, and the information is compressed by quantizing each two-dimensional block regarding the time and frequency. , Recording or transmitting together with the information compression parameter for each two-dimensional block relating to the time and frequency, and / or expressing the signal component in the plurality of two-dimensional blocks relating to the time and frequency for which information compression has been performed, In a compressed data recording and / or reproducing or transmitting and / or receiving method of reproducing or receiving using information compression parameters for each block, a mode of recording and / or reproducing of plural information bit rates or a mode of transmitting and / or receiving And the number of information compression parameters depends on the information bit rate of each mode. Compressed data and wherein the varying recording and /
Or a reproduction or transmission and / or reception method. 37. A mode having a higher information bit rate,
Record or transmit more information parameters, and / or
Alternatively, it is played back or received.
6. The compressed data recording and / or reproducing or transmitting and / or receiving method according to 6. 38. A mode in which the information bit rate is lower,
Record or transmit fewer information parameters, and / or
Alternatively, it is played back or received.
6. The compressed data recording and / or reproducing or transmitting and / or receiving method according to 6. 39. The method for recording and / or reproducing or transmitting and / or receiving compressed data according to claim 37, wherein the number of information compression parameters changes in proportion to the information bit rate of each mode. 40. By collectively recording or transmitting and / or reproducing or receiving information compression parameters of at least two two-dimensional blocks arranged in the time direction or the frequency direction or the time and frequency directions, the information compression parameters can be changed. 40. A method of recording and / or reproducing or transmitting and / or receiving compressed data according to claim 39, characterized in that a number reduction is achieved. 41. Information compression by collectively recording or transmitting and / or reproducing or receiving information compression parameters of at least two adjacent two-dimensional blocks arranged in the time direction or frequency direction or time and frequency direction. 40. A method of recording and / or reproducing or transmitting and / or receiving compressed data according to claim 39, characterized in that a reduction in the number of parameters is achieved. 42. A mode in which the information bit rate is lower,
42. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 40, 41, wherein a larger number of information compression parameters are collectively recorded or transmitted and / or reproduced or received. 43. The information compression parameter according to claim 42, wherein the number of information compression parameters to be grouped is made inversely proportional to the information bit rate for each mode, and a larger number of information compression parameters are grouped in a mode having a lower information bit rate. Compressed data recording and / or reproducing or transmitting and / or receiving method. 44. The compressed data recording and / or reproducing or transmitting and / or receiving according to claim 43, wherein the two-dimensional block is a block for block floating and / or a quantization noise generation control block. Method. 45. The input signal is an audio signal, and the frequency width of at least most of the quantization noise generation control blocks is made wider in a higher frequency range.
5. A method of recording and / or reproducing or transmitting and / or receiving compressed data according to 5,44. 46. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 45, wherein the sampling frequencies of all modes are the same. 47. Using an orthogonal transform to decompose a digital signal into a plurality of frequency components to obtain a signal within a two-dimensional block of time and frequency, and / or within a two-dimensional block of time and frequency. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 35 or 44, characterized in that an inverse orthogonal transform is used for converting the signal into a digital signal on a time axis. 48. To decompose a digital signal into a plurality of frequency band components and obtain signal components in a plurality of two-dimensional blocks regarding time and frequency, first divide the digital signal into a plurality of bands, and for each of the divided bands. For each block of each band, a block consisting of multiple samples is formed, coefficient data is obtained by performing orthogonal transform for each block of each band, and / or conversion from multiple bands on the frequency axis to time axis signals is performed. 48. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 47, wherein inverse orthogonal transformation is performed and respective inverse orthogonal transformation outputs are combined to obtain a combined signal on a time axis. 49. The block size of the orthogonal transform and / or the inverse orthogonal transform is variable.
48. A compressed data recording and / or reproducing or transmitting and / or receiving method according to item 48. 50. A division frequency width in dividing a time axis signal before orthogonal transformation into a plurality of bands on a frequency axis and / or a combination of a plurality of bands on a frequency axis after inverse orthogonal transformation into a time axis signal Combined frequency width from multiple bands,
50. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 48 or 49, characterized in that two consecutive lowest bands are the same. 51. A division frequency width in dividing a time axis signal before orthogonal transformation into a plurality of bands on a frequency axis and / or a combination of a plurality of bands on a frequency axis after inverse orthogonal transformation into a time axis signal Combined frequency width from multiple bands,
51. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 50, characterized in that the widening is made approximately as high as possible. 52. The compressed data recording and / or reproducing or transmitting and / or receiving according to claim 51, wherein a modified discrete cosine transform is used as the orthogonal transform and an inverse modified discrete cosine transform is used as the inverse orthogonal transform. Method. 53. Compressed data recording and / or reproducing or transmitting and / or receiving according to claim 51, wherein the transform block size of orthogonal transform is the same between all modes having different information bit rates. Method. 54. When converting compressed data between modes having different information bit rates, the compressed data is converted with the signal component on the frequency axis as it is without returning to the signal component on the time axis. 54. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 53. 55. When converting compressed data from a mode having a higher information bit rate to a mode having a lower information bit rate, the signal component on the frequency axis remains as it is without returning to the signal component on the time axis. 55. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 54, characterized in that the compressed data is converted by. 56. When conversion of compressed data from a mode having a lower information bit rate to a mode having a higher information bit rate is performed, the encoded compressed data is not decoded but is in an encoded state. The compressed data recording and / or reproducing or transmitting and / or receiving method according to claim 55, wherein the compressed data is converted as it is. 57. A recording medium on which compressed data recorded by the compressed data recording apparatus according to claim 1, 8, 19 or 20 is recorded. 58. The recording medium according to claim 57, comprising a magneto-optical disk recording medium. 59. The recording medium according to claim 57, comprising a semiconductor recording medium. 60. The recording medium according to claim 57, comprising an IC memory card recording medium. 61. The recording medium according to claim 57, comprising an optical disc recording medium.